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Feat/3pcc invite (#187)

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* wip

* wip

* add test for late media / 3pcc invite, which should now work
This commit is contained in:
Dave Horton
2024-12-12 18:39:01 -05:00
committed by GitHub
parent a7d6cf5a31
commit c807fdb93d
3 changed files with 90 additions and 27 deletions
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77
test/scenarios/uac-late-media.xml
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@@ -19,22 +19,21 @@
<!-- Sipp 'uac' scenario with pcap (rtp) play -->
<!-- -->
<scenario name="UAC with late media">
<scenario name="UAC with media">
<!-- In client mode (sipp placing calls), the Call-ID MUST be -->
<!-- generated by sipp. To do so, use [call_id] keyword. -->
<send retrans="500">
<![CDATA[
INVITE sip:16173333456@[remote_ip]:[remote_port] SIP/2.0
INVITE sip:+16173333456@jambonz.org SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[call_number]
To: sut <sip:[service]@[remote_ip]:[remote_port]>
From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[pid]SIPpTag09[call_number]
To: <sip:16173333456@jambonz.org>
Call-ID: [call_id]
CSeq: 1 INVITE
Contact: sip:sipp@[local_ip]:[local_port]
Max-Forwards: 70
Subject: uac-no-3pcc
Content-Type: application/sdp
Subject: uac-late-media
Content-Length: 0
]]>
@@ -43,27 +42,79 @@
<recv response="100" optional="true">
</recv>
<recv response="488">
<recv response="180" optional="true">
</recv>
<!-- By adding rrs="true" (Record Route Sets), the route sets -->
<!-- are saved and used for following messages sent. Useful to test -->
<!-- against stateful SIP proxies/B2BUAs. -->
<recv response="200" rtd="true" crlf="true">
</recv>
<!-- Packet lost can be simulated in any send/recv message by -->
<!-- by adding the 'lost = "10"'. Value can be [1-100] percent. -->
<send>
<![CDATA[
ACK sip:16173333456@[remote_ip]:[remote_port] SIP/2.0
[last_Via]
ACK sip:16173333456@jambonz.org SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[pid]SIPpTag09[call_number]
To: [service] <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
To: <sip:16173333456@jambonz.org>[peer_tag_param]
Call-ID: [call_id]
CSeq: 1 ACK
Subject: uac-no-3pcc
Max-Forwards: 70
Subject: uac-late-media
Content-Type: application/sdp
Content-Length: [len]
v=0
o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
s=-
c=IN IP[local_ip_type] [local_ip]
t=0 0
m=audio [auto_media_port] RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11,16
]]>
</send>
<!-- Play a pre-recorded PCAP file (RTP stream) -->
<nop>
<action>
<exec play_pcap_audio="pcap/g711a.pcap"/>
</action>
</nop>
<!-- Pause briefly -->
<pause milliseconds="3000"/>
<!-- The 'crlf' option inserts a blank line in the statistics report. -->
<send retrans="500">
<![CDATA[
BYE sip:16173333456@jambonz.org SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[pid]SIPpTag09[call_number]
To: <sip:16173333456@jambonz.org>[peer_tag_param]
Call-ID: [call_id]
CSeq: 2 BYE
Subject: uac-late-media
Content-Length: 0
]]>
</send>
<!-- definition of the response time repartition table (unit is ms) -->
<recv response="200" crlf="true">
</recv>
<!-- definition of the response time repartition table (unit is ms) -->
<ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/>
<!-- definition of the call length repartition table (unit is ms) -->
<CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/>
</scenario>
</scenario>